Asterisk 11.6 + Sipml5

aleksandr_star

Добрый день.

Настроил работу Asterisk 11.6 + Sipml5 в локальной сети исходящие работают голос есть. Проблема заключается в том, что при использовании Asterisk и Sipml5 которые находятся на одном сервере с белым ip-адресом и самими сотрудники находятся за NAT-ом голос не проходит. Stun сервер использую stun.l.google.com:19302, так же пробовал подымать свой stun, результат не изменился. При выполнении rtp set debug on видно что идут пакеты только "Sent RTP packet to" непонятно почему нет пакетов "Got RTP packet from".

ded · Сообщение **ded** » 11 май 2017, 11:18

Если идут пакеты только "Sent RTP packet to" и нет пакетов "Got RTP packet from", то это потому, что в пакетах INVITE, которые инициализировали это соединение, указан неверный ИП адрес Destination для медиа. Чтобы это понять/простить - нужно
включить sip debug
отловить INVITE и ответ на него !
разобраться наконец.

Мы используем stun.sipgate.net и он не подводил пока.

aleksandr_star

IP-адрес в сообщениях Invite указан наш внешний. Для примера прикрепил лог звонка:
185.13.249.85 - офис с сотрудниками
95.213.132.131 - сервер
92 - эхо тест

PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

<--- SIP read from WS:185.13.249.85:52074 --->
INVITE sip:92@95.213.132.131 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKeRpElKs2yv4BRFmBWD2cf21PQWX9TnAL;rport
From: "sipml5001"<sip:sipml5001@95.213.132.131:5084>;tag=sUwPB6JnGa0GVId0N7I4
To: <sip:92@95.213.132.131>
Contact: "sipml5001"<sip:sipml5001@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=sipml5001;ha1=4aaf669a311a32dea25e644113e8f4bc;+g.oma.sip-im;language="en,fr"
Call-ID: ff4857cc-64f2-cdfe-7fd8-ddeda7e0842d
CSeq: 40134 INVITE
Content-Type: application/sdp
Content-Length: 1699
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom

v=0
o=mozilla...THIS_IS_SDPARTA-53.0 4681333663516931000 0 IN IP4 127.0.0.1
s=Doubango Telecom - firefox
t=0 0
a=sendrecv
a=fingerprint:sha-256 9A:95:60:7B:EC:DD:4F:98:45:BE:B8:F0:75:FC:5F:78:67:EA:DB:60:58:7F:AE:ED:83:48:56:90:39:A4:2D:8C
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 36802 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 185.13.249.85
a=candidate:0 1 UDP 2122187007 192.168.0.21 36802 typ host
a=candidate:2 1 UDP 2122252543 192.168.1.51 47133 typ host
a=candidate:4 1 UDP 2122121471 192.168.122.1 49802 typ host
a=candidate:0 2 UDP 2122187006 192.168.0.21 41764 typ host
a=candidate:2 2 UDP 2122252542 192.168.1.51 40857 typ host
a=candidate:4 2 UDP 2122121470 192.168.122.1 52144 typ host
a=candidate:1 1 UDP 1685987327 185.13.249.85 36802 typ srflx raddr 192.168.0.21 rport 36802
a=candidate:5 1 UDP 1685921791 185.13.249.85 49802 typ srflx raddr 192.168.122.1 rport 49802
a=candidate:1 2 UDP 1685987326 185.13.249.85 41764 typ srflx raddr 192.168.0.21 rport 41764
a=candidate:5 2 UDP 1685921790 185.13.249.85 52144 typ srflx raddr 192.168.122.1 rport 52144
a=sendrecv
a=end-of-candidates
a=extmap:1/sendonly urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:705adaf5743b7b267689f1d4cbef53e6
a=ice-ufrag:903ec4a4
a=mid:sdparta_0
a=msid:{715cf9c2-fd27-4014-8221-a2c2bcb16442} {ffd1e8b5-7d6d-4bca-8d3c-bf95b6a6109c}
a=rtcp:41764 IN IP4 185.13.249.85
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:2576317965 cname:{5f58851e-006f-4a20-b1c9-f7d4f624a2d2}
<------------->
--- (12 headers 38 lines) ---
Using INVITE request as basis request - ff4857cc-64f2-cdfe-7fd8-ddeda7e0842d
Found peer 'sipml5001' for 'sipml5001' from 185.13.249.85:52074

<--- Reliably Transmitting (NAT) to 185.13.249.85:52074 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKeRpElKs2yv4BRFmBWD2cf21PQWX9TnAL;received=185.13.249.85;rport=52074
From: "sipml5001"<sip:sipml5001@95.213.132.131:5084>;tag=sUwPB6JnGa0GVId0N7I4
To: <sip:92@95.213.132.131>;tag=as12e4ae63
Call-ID: ff4857cc-64f2-cdfe-7fd8-ddeda7e0842d
CSeq: 40134 INVITE
Server: Phonet
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="190", nonce="6e249fea"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'ff4857cc-64f2-cdfe-7fd8-ddeda7e0842d' in 7488 ms (Method: INVITE)

<--- SIP read from WS:185.13.249.85:52074 --->
ACK sip:92@95.213.132.131 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKeRpElKs2yv4BRFmBWD2cf21PQWX9TnAL;rport
From: "sipml5001"<sip:sipml5001@95.213.132.131:5084>;tag=sUwPB6JnGa0GVId0N7I4
To: <sip:92@95.213.132.131>;tag=as12e4ae63
Call-ID: ff4857cc-64f2-cdfe-7fd8-ddeda7e0842d
CSeq: 40134 ACK
Content-Length: 0
Max-Forwards: 70

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from WS:185.13.249.85:52074 --->
INVITE sip:92@95.213.132.131 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKGJJaInHCKyxfzG1TbepzTBnTpFJPH601;rport
From: "sipml5001"<sip:sipml5001@95.213.132.131:5084>;tag=sUwPB6JnGa0GVId0N7I4
To: <sip:92@95.213.132.131>
Contact: "sipml5001"<sip:sipml5001@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;impi=sipml5001;ha1=4aaf669a311a32dea25e644113e8f4bc;+g.oma.sip-im;language="en,fr"
Call-ID: ff4857cc-64f2-cdfe-7fd8-ddeda7e0842d
CSeq: 40135 INVITE
Content-Type: application/sdp
Content-Length: 1699
Max-Forwards: 70
Authorization: Digest username="sipml5001",realm="190",nonce="6e249fea",uri="sip:92@95.213.132.131",response="20ef4c6770a1bd10dfeeb2150514c351",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom

v=0
o=mozilla...THIS_IS_SDPARTA-53.0 4681333663516931000 0 IN IP4 127.0.0.1
s=Doubango Telecom - firefox
t=0 0
a=sendrecv
a=fingerprint:sha-256 9A:95:60:7B:EC:DD:4F:98:45:BE:B8:F0:75:FC:5F:78:67:EA:DB:60:58:7F:AE:ED:83:48:56:90:39:A4:2D:8C
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 36802 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 185.13.249.85
a=candidate:0 1 UDP 2122187007 192.168.0.21 36802 typ host
a=candidate:2 1 UDP 2122252543 192.168.1.51 47133 typ host
a=candidate:4 1 UDP 2122121471 192.168.122.1 49802 typ host
a=candidate:0 2 UDP 2122187006 192.168.0.21 41764 typ host
a=candidate:2 2 UDP 2122252542 192.168.1.51 40857 typ host
a=candidate:4 2 UDP 2122121470 192.168.122.1 52144 typ host
a=candidate:1 1 UDP 1685987327 185.13.249.85 36802 typ srflx raddr 192.168.0.21 rport 36802
a=candidate:5 1 UDP 1685921791 185.13.249.85 49802 typ srflx raddr 192.168.122.1 rport 49802
a=candidate:1 2 UDP 1685987326 185.13.249.85 41764 typ srflx raddr 192.168.0.21 rport 41764
a=candidate:5 2 UDP 1685921790 185.13.249.85 52144 typ srflx raddr 192.168.122.1 rport 52144
a=sendrecv
a=end-of-candidates
a=extmap:1/sendonly urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:705adaf5743b7b267689f1d4cbef53e6
a=ice-ufrag:903ec4a4
a=mid:sdparta_0
a=msid:{715cf9c2-fd27-4014-8221-a2c2bcb16442} {ffd1e8b5-7d6d-4bca-8d3c-bf95b6a6109c}
a=rtcp:41764 IN IP4 185.13.249.85
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:2576317965 cname:{5f58851e-006f-4a20-b1c9-f7d4f624a2d2}
<------------->
--- (13 headers 38 lines) ---
Using INVITE request as basis request - ff4857cc-64f2-cdfe-7fd8-ddeda7e0842d
Found peer 'sipml5001' for 'sipml5001' from 185.13.249.85:52074
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Found RTP audio format 109
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found unknown media description format opus for ID 109
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw|alaw|g722)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 185.13.249.85:36802
Looking for 92 in default (domain 95.213.132.131)
list_route: hop: <sip:sipml5001@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>

<--- Transmitting (NAT) to 185.13.249.85:52074 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKGJJaInHCKyxfzG1TbepzTBnTpFJPH601;received=185.13.249.85;rport=52074
From: "sipml5001"<sip:sipml5001@95.213.132.131:5084>;tag=sUwPB6JnGa0GVId0N7I4
To: <sip:92@95.213.132.131>
Call-ID: ff4857cc-64f2-cdfe-7fd8-ddeda7e0842d
CSeq: 40135 INVITE
Server: Phonet
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:92@95.213.132.131:5084;transport=WS>
Content-Length: 0

<------------>
-- Executing [92@default:1] Verbose("SIP/sipml5001-00000000", "5, default.92: Echo test for "sipml5001" <sipml5001>") in new stack
> default.92: Echo test for "sipml5001" <sipml5001>
-- Executing [92@default:2] Answer("SIP/sipml5001-00000000", "") in new stack
Audio is at 35564
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 185.13.249.85:52074 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKGJJaInHCKyxfzG1TbepzTBnTpFJPH601;received=185.13.249.85;rport=52074
From: "sipml5001"<sip:sipml5001@95.213.132.131:5084>;tag=sUwPB6JnGa0GVId0N7I4
To: <sip:92@95.213.132.131>;tag=as54792285
Call-ID: ff4857cc-64f2-cdfe-7fd8-ddeda7e0842d
CSeq: 40135 INVITE
Server: Phonet
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:92@95.213.132.131:5084;transport=WS>
Content-Type: application/sdp
Content-Length: 1580

v=0
o=Phonet 1854440499 1854440499 IN IP4 95.213.132.131
s=Phonet
c=IN IP4 95.213.132.131
t=0 0
m=audio 35564 RTP/SAVPF 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:6527545e6bbc36d92d293e166efd1dc4
a=ice-pwd:1def5b9f4cd2bdd8109c34c003c244e2
a=candidate:H5fd58483 1 UDP 2130706431 95.213.132.131 35564 typ host
a=candidate:Hbc5d145a 1 UDP 2130706431 188.93.20.90 35564 typ host
a=candidate:H5fd58484 1 UDP 2130706431 95.213.132.132 35564 typ host
a=candidate:H5fd58485 1 UDP 2130706431 95.213.132.133 35564 typ host
a=candidate:H5fd58486 1 UDP 2130706431 95.213.132.134 35564 typ host
a=candidate:Hbc5d145b 1 UDP 2130706431 188.93.20.91 35564 typ host
a=candidate:Hbc5d145c 1 UDP 2130706431 188.93.20.92 35564 typ host
a=candidate:Hbc5d145d 1 UDP 2130706431 188.93.20.93 35564 typ host
a=candidate:Hbc5d145e 1 UDP 2130706431 188.93.20.94 35564 typ host
a=candidate:S5fd58483 1 UDP 1694498815 95.213.132.131 35564 typ srflx
a=candidate:H5fd58483 2 UDP 2130706430 95.213.132.131 35565 typ host
a=candidate:Hbc5d145a 2 UDP 2130706430 188.93.20.90 35565 typ host
a=candidate:H5fd58484 2 UDP 2130706430 95.213.132.132 35565 typ host
a=candidate:H5fd58485 2 UDP 2130706430 95.213.132.133 35565 typ host
a=candidate:H5fd58486 2 UDP 2130706430 95.213.132.134 35565 typ host
a=candidate:Hbc5d145b 2 UDP 2130706430 188.93.20.91 35565 typ host
a=connection:new
a=setup:active
a=fingerprint:SHA-256 EE:6D:C1:3B:5C:84:94:6D:51:67:4E:74:84:ED:B7:0B:B7:2C:B3:A7:4F:90:55:25:E0:7F:78:BC:E7:56:AF:63
a=sendrecv

<------------>

<--- SIP read from WS:185.13.249.85:52074 --->
ACK sip:92@95.213.132.131:5084;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKGlB5C68z705MxfTywtq2;rport
From: "sipml5001"<sip:sipml5001@95.213.132.131:5084>;tag=sUwPB6JnGa0GVId0N7I4
To: <sip:92@95.213.132.131>;tag=as54792285
Contact: "sipml5001"<sip:sipml5001@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws>;+g.oma.sip-im;language="en,fr"
Call-ID: ff4857cc-64f2-cdfe-7fd8-ddeda7e0842d
CSeq: 40135 ACK
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="sipml5001",realm="190",nonce="6e249fea",uri="sip:92@95.213.132.131:5084;transport=WS",response="d955107f22bb54aeeaf77fbeb26037ae",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom

<------------->
--- (12 headers 0 lines) ---
-- Executing [92@default:3] Playback("SIP/sipml5001-00000000", "demo-echotest") in new stack
Sent RTP packet to 185.13.249.85:36802 (type 00, seq 057370, ts 000160, len 000160)
-- <SIP/sipml5001-00000000> Playing 'demo-echotest.ulaw' (language 'ru')
Sent RTP packet to 185.13.249.85:36802 (type 00, seq 057371, ts 000320, len 000160)

virus_net · Сообщение **virus_net** » 11 май 2017, 12:34

Peer audio RTP is at port 185.13.249.85:36802

c=IN IP4 95.213.132.131
m=audio 35564

Соответственно в первую очередь смотрите есть ли трафик от 192.168.0.21 к 95.213.132.131 и от 185.13.249.85 к 95.213.132.131

ded · Сообщение **ded** » 11 май 2017, 12:43

Если 95.213.132.131 - сервер, то порт m=audio 35564 - подозрителен. Обычно диапазон портов для медиа 10000-20000 (файл rtp.conf)

aleksandr_star

у меня просто rtp порты немного другие
rtpstart=34000
rtpend=35998

ded · Сообщение **ded** » 11 май 2017, 12:57

OK, то есть Астериск понимает, что Peer audio RTP is at port 185.13.249.85:36802, но ничего туда не отправляет? Судя по логу он устанавливает ulaw для этого соединения, а значит всё должно уходить.

aleksandr_star

Да конечно понимает, так как простой софтфон или же телефонный аппарат работает

aleksandr_star

virus_net писал(а):
Peer audio RTP is at port 185.13.249.85:36802

c=IN IP4 95.213.132.131
m=audio 35564
Соответственно в первую очередь смотрите есть ли трафик от 192.168.0.21 к 95.213.132.131 и от 185.13.249.85 к 95.213.132.131

Попробую проверить

aleksandr_star

virus_net писал(а):
Peer audio RTP is at port 185.13.249.85:36802

c=IN IP4 95.213.132.131
m=audio 35564
Соответственно в первую очередь смотрите есть ли трафик от 192.168.0.21 к 95.213.132.131 и от 185.13.249.85 к 95.213.132.131

Проверил. Пакеты идут только в одну сторону http://ipic.su/img/img7/fs/Snimokekrana ... 513049.png (верхний скрин - компьютер где работает Sipml5 и нижний запущенно было на сервере с АТС)

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